Add "autoframing" option to sip.conf.sample and h323.conf.sample.

The autoframing option was added to chan_sip.c in r43243 (mogorman,
2006-09-19 01:32:57), but never made its way into the sample configs.

Review: https://reviewboard.asterisk.org/r/2768/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@396994 65c4cc65-6c06-0410-ace0-fbb531ad65f3

configs/h323.conf.sample

@@ -29,6 +29,8 @@ port = 1720
 ;allow=gsm		; Always allow GSM, it's cool :)
 ;allow=ulaw		; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
 			; for framing options
+;autoframing=yes	; Set packetization based on the remote endpoint's (ptime)
+			; preferences. Defaults to no.
 ;
 ; User-Input Mode (DTMF)
 ;

configs/sip.conf.sample

@@ -300,6 +300,8 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;allow=ulaw                     ; Allow codecs in order of preference
 ;allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
 				; for framing options
+;autoframing=yes		; Set packetization based on the remote endpoint's (ptime)
+				; preferences. Defaults to no.
 ;
 ; This option specifies a preference for which music on hold class this channel
 ; should listen to when put on hold if the music class has not been set on the
@@ -1121,6 +1123,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; language
 ; allow
 ; disallow
+; autoframing
 ; insecure
 ; trustrpid
 ; progressinband